Abstract: A speech coding apparatus compares the closeness of the feature value of a feature vector signal of an utterance to the parameter values of prototype vector signals to obtain prototype match scores for the feature vector signal and each prototype vector signal. The speech coding apparatus stores a plurality of speech transition models representing speech transitions. At least one speech transition is represented by a plurality of different models. Each speech transition model has a plurality of model outputs, each comprising a prototype match score for a prototype vector signal. Each model output has an output probability. A model match score for a first feature vector signal and each speech transition model comprises the output probability for at least one prototype match score for the first feature vector signal and a prototype vector signal.

Abstract: A dictionary order sorter resorts character strings of recognition candidates stored in a high-ranking candidate memory in the order of distance into a dictionary order (character code order). Upon receipt of a sort termination signal a display controller displays the character strings of recognition candidates stored in the high-ranking candidate memory in the dictionary order and their ranking numbers in order on a display. Where an attribute-dependent sorter is provided in place of the dictionary order sorter, the character strings of recognition candidates stored in the distance order in the high-ranking candidate memory are sorted (grouped) according to attribute information of categories, such as parts of speech, concepts, etc., which are stored in a template memory for each of the recognition candidates and then displayed classified into groups. The categories of attribute information can arbitrarily be specified by the user.

Abstract: Speech coding of the code excited linear predictive type is implemented by providing an excitation vector which comprises a set of a pre-determined number of pulse patterns from a codebook of P pulse patterns, which have a selected orientation and a pre-determined delay with respect to the starting point of the excitation vector. This requires modest computational power and a small memory space, which allows it to be implemented in one signal processor.

Abstract: A code excited linear predictive coder and decoder well suited to speech recording, transmission and reproduction, especially in voice messaging systems, provides backward adaptive gain control of stored codevectors to be applied to a synthesis filter prior to being compared with sequences of input speech signals. Simplified linear predictive parameter quantization using efficient table lookup procedures, efficient codevector storage and search all contribute in an illustrative embodiment to high quality coding and decoding with reduced computational complexity.

Abstract: A method and apparatus for the automatic analysis, synthesis and modification of audio signals, based on an overlap-add sinusoidal model, is disclosed. Automatic analysis of amplitude, frequency and phase parameters of the model is achieved using an analysis-by-synthesis procedure which incorporates successive approximation, yielding synthetic waveforms which are very good approximations to the original waveforms and are perceptually identical to the original sounds. A generalized overlap-add sinusoidal model is introduced which can modify audio signals without objectionable artifacts. In addition, a new approach to pitch-scale modification allows for the use of arbitrary spectral envelope estimates and addresses the problems of high-frequency loss and noise amplification encountered with prior art methods.

Abstract: A speech signal coding apparatus inputs a pitch period generated by coding a speech signal, and outputs information on a range for said pitch period, together with the pitch period. A speech signal decoding apparatus inputs the pitch period and the above information on the range, and determines whether or not the pitch period is within the range. When the pitch period is determined to be within the range, the speech signal decoding apparatus outputs the above pitch period. When the pitch period is determined not to be within the range, the speech signal decoding apparatus outputs as a pitch period a predetermined value within the range.

Abstract: A speech coding system is provided where input speech is coded by finding via an evaluation computation a code vector giving a minimum error between reproduced signals obtained by linear prediction analysis filter processing, simulating speech path characteristics, on code vectors successively read out from a noise codebook storing a plurality of noise trains as code vectors and an input speech signal and by using a code specifying the code vector. In the speech coding system, the noise codebook includes a delta vector codebook which stores an initial vector and a plurality of delta vectors having difference vectors between adjoining code vectors. In addition, provision is made in the computing unit for the evaluation computation of a cyclic adding unit for cumulatively adding the delta vectors to virtually reproduce the code vectors.

Abstract: A voice synthesizing apparatus is arranged to synthesize a voice from text data composed of either character codes or a series of symbols by generating a sound source based on a series of sound-source parameters and synthesizing the sound source on the basis of a series of synthesis parameters. The voice synthesizing apparatus is provided with a sound-source generating circuit for generating the aforesaid sound source from a signal obtained from an instrumental sound generated with a musical instrument. This arrangement serves to easily synthesize voices which convey language information and yet which simulate the sounds of musical instruments such as a guitar, a violin, a harmonica, a musical synthesizer and the like.

Abstract: A method and apparatus for allocating transmission bits for use in transmitting samples of a digital signal. An aggregate allowable quantization distortion value is selected representing an allowable quantization distortion error for a frame of samples of the digital signal. A set of samples are selected from the frame of samples such that a plurality of the selected samples are greater than a noise threshold. For each sample of the set, a sample quantization distortion value is computed which represents an allowable quantization distortion error for the sample. The sum of all sample quantization distortion values is approximately equal to the aggregate allowable quantization distortion value. For each sample of the set, a quantization step size is selected which yields a quantization distortion error approximately equal to the sample's corresponding quantization distortion value. Each sample is then quantized using its quantization step size.

Abstract: A speech categorization system includes first and second timers which generate first and second measured durations indicative of duration of selected higher and lower amplitude segments included in a voice message. A higher amplitude segment is classified in a first category when the first and second measured durations corresponding to the higher amplitude segment and an adjacent lower amplitude segment satisfy a classification test, and a counter counts the number of the higher amplitude segments classified in the first category Accented syllables in the higher amplitude segment are recognized to aid classification.

Abstract: A speech recognition system includes a parameter extracting section for extracting a speech parameter of input speech, a first recognizing section for performing recognition processing by word-based matching, and a second recognizing section for performing word recognition by matching in units of word constituent elements. The first word recognizing section segments the speech parameter in units of words to extract a word speech pattern and performs word recognition by matching the word speech pattern with a predetermined word reference pattern. The second word recognizing section performs recognition in units of word constituent elements by using the extracted speech parameter and performs word recognition on the basis of candidates of an obtained word constituent element series.

Abstract: Signal processing structures for providing direct prediction coefficients and direct Least Square-Finite Impulse Response (LS-FIR) filter coefficients. The structures include one or more processors, and a storage and retrieval structure for selectively storing predictor and filter coefficients and intermediate variables, to thereby allow the one or more real processors to emulate a plurality of virtual processors, which take the form of a side fed superlattice structure, in the case of linear prediction, and a side-fed superlattice-superladder structure, in the case of direct LS-FIR filtering.

Abstract: A quantizer, which converts an input group of data samples into one of N quantized groups of data samples, does so by performing several dot products and compares. Each dot product is between the input group of data samples and a reference group of data samples u.sub.x, and each compare is with the dot product result and a constant k.sub.y. All of the reference groups of data samples and all of the constants are stored in a memory. This memory holds only N-1 constants and less then N/2 reference groups of data samples. A memory saving of several thousand percent is achieved by limiting the memory to hold no more then 2r reference groups of data samples where r is the number of samples in the input group, or no more then 2 log.sub.2 N reference groups of data samples.

Abstract: A Lombard voice recognition method for recognizing a voice input in a noisy background includes a step of matching and a step of warping. The step of matching matches a frequency spectrum of a feature pattern derived from the input voice to a frequency spectrum of a standard pattern. The step of warping warps a lower frequency spectrum of the feature pattern than a lower frequency side with respect to a frequency axis in case that a noise level of the background in higher than a predetermined noise level.

Abstract: An exemplary CELP coder where gain adaptation is performed using previous gain values in conjunction with an entry in a table comprising the logarithms of the root-mean-squared values of the codebook vectors, to predict the next gain value. Not only is this method less complex because the table entries are determined off-line, but in addition the use of a table at both the encoder and the decoder allows fixed-point/floating-point interoperability requirements to be met.

Abstract: Input speech of a reference speaker, who wants to convert his/her voice quality, and speech of a target speaker are converted into a digital signal by an analog to digital (A/D) converter. The digital signal is then subjected to speech analysis by a linear predictive coding (LPC) analyzer. Speech data of the reference speaker is processed into speech segments by a speech segmentation unit. A speech segment correspondence unit makes a dynamic programming (DP) based correspondence between the obtained speech segments and training speech data of the target speaker, thereby making a speech segment correspondence table. A speaker individuality conversion is made on the basis of the speech segment correspondence table by a speech individuality conversion and synthesis unit.

Abstract: A new basis vector search process that directly results in an optimal linear weighting for a VSELP (Vector Sum Excited Linear Prediction) coder, thus avoiding the need to perform an extensive search. In the present invention, the conventional search process is replaced by a direct formula, thus avoiding the time consuming searching procedure. Using a simple mathematical relationship, the process of filtering the basis signals with an impulse response filter h(n) every subframe is avoided. A simple theorem has been developed to reduce the computation involved in carrying out the filtering of the basis signals with h(n), and is referred to as the switching convolution theorem. As a result, the computation time necessary to produce the optimal weighting is reduced by a factor of from 3 to 4, while maintaining the output quality of the coder. The new apparatus and method are based upon a set of equations that includes several experimentally justified assumptions.

Abstract: A speech codec operating at low data rates uses an iterative method to jointly optimize pitch and gain parameter sets. A 26-bit spectrum filter coding scheme may be used, involving successive subtractions and quantizations. The codec may preferably use a decomposed multipulse excitation model, wherein the multipulse vectors used as the excitation signal are decomposed into position and amplitude codewords. Multipulse vectors are coded by comparing each vector to a reference multipulse vector and quantizing the resulting difference vector. An expanded multipulse excitation codebook and associated fast search method, optionally with a dynamically-weighted distortion measure, allow selection of the best excitation vector without memory or computational overload. In a dynamic bit allocation technique, the number of bits allocated to the pitch and excitation signals depend on whether the signals are "significant" or "insignificant".

Abstract: A method and an apparatus for hearing assistance, capable of compensating the lowering of the speech recognition ability related to the deterioration of the auditory sense center. The input speech is divided into voiced speech sections, unvoiced speech sections, and silent sections, of which the voiced speech sections and the silent sections are appropriately extended/contracted while the unvoiced speech sections are left unchanged, and then these sections are combined in an identical order as in the input speech, so as to obtain output speech which is easier to listen for a listener with a handicapped hearing ability.

Abstract: A speech coder apparatus operates to compress speech signals to a low bit rate. The apparatus includes a continuous speech recognizer (CSR) which has a memory for storing templates. Input speech is processed by the CSR where information in the speech is compared against the templates to provide an output digital signal indicative of recognized words, which signal is transmitted along a first path. There is further included a front end processor which is also responsive to the input speech signal for providing output digitized speech samples during a given frame interval. A side information encoder circuit responds to the output from the front end processor to provide at the output of the encoder a parameter signal indicative of the value of the pitch and word duration for each word as recognized by the CSR unit. The output of the encoder is transmitted as a second signal.